This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
196 lines
6.0 KiB
C++
196 lines
6.0 KiB
C++
/*
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* Copyright 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <memory>
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#include "api/test/create_frame_generator.h"
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#include "api/test/frame_generator_interface.h"
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#include "api/test/simulated_network.h"
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#include "call/fake_network_pipe.h"
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#include "call/simulated_network.h"
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#include "rtc_base/task_queue_for_test.h"
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#include "system_wrappers/include/sleep.h"
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#include "test/call_test.h"
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#include "test/field_trial.h"
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#include "test/frame_forwarder.h"
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#include "test/gtest.h"
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#include "test/null_transport.h"
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namespace webrtc {
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class CallOperationEndToEndTest : public test::CallTest {};
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TEST_F(CallOperationEndToEndTest, ReceiverCanBeStartedTwice) {
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CreateCalls();
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test::NullTransport transport;
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CreateSendConfig(1, 0, 0, &transport);
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CreateMatchingReceiveConfigs(&transport);
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CreateVideoStreams();
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video_receive_streams_[0]->Start();
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video_receive_streams_[0]->Start();
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DestroyStreams();
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}
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TEST_F(CallOperationEndToEndTest, ReceiverCanBeStoppedTwice) {
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CreateCalls();
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test::NullTransport transport;
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CreateSendConfig(1, 0, 0, &transport);
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CreateMatchingReceiveConfigs(&transport);
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CreateVideoStreams();
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video_receive_streams_[0]->Stop();
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video_receive_streams_[0]->Stop();
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DestroyStreams();
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}
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TEST_F(CallOperationEndToEndTest, ReceiverCanBeStoppedAndRestarted) {
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CreateCalls();
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test::NullTransport transport;
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CreateSendConfig(1, 0, 0, &transport);
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CreateMatchingReceiveConfigs(&transport);
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CreateVideoStreams();
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video_receive_streams_[0]->Stop();
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video_receive_streams_[0]->Start();
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video_receive_streams_[0]->Stop();
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DestroyStreams();
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}
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TEST_F(CallOperationEndToEndTest, RendersSingleDelayedFrame) {
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static const int kWidth = 320;
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static const int kHeight = 240;
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// This constant is chosen to be higher than the timeout in the video_render
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// module. This makes sure that frames aren't dropped if there are no other
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// frames in the queue.
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static const int kRenderDelayMs = 1000;
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class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
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public:
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void OnFrame(const VideoFrame& video_frame) override {
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SleepMs(kRenderDelayMs);
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event_.Set();
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}
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bool Wait() { return event_.Wait(kDefaultTimeout); }
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rtc::Event event_;
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} renderer;
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test::FrameForwarder frame_forwarder;
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SendTask(
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task_queue(), [this, &renderer, &frame_forwarder]() {
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CreateCalls();
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CreateSendTransport(BuiltInNetworkBehaviorConfig(),
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/*observer=*/nullptr);
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CreateReceiveTransport(BuiltInNetworkBehaviorConfig(),
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/*observer=*/nullptr);
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CreateSendConfig(1, 0, 0);
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CreateMatchingReceiveConfigs();
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video_receive_configs_[0].renderer = &renderer;
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CreateVideoStreams();
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Start();
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// Create frames that are smaller than the send width/height, this is
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// done to check that the callbacks are done after processing video.
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std::unique_ptr<test::FrameGeneratorInterface> frame_generator(
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test::CreateSquareFrameGenerator(kWidth, kHeight, absl::nullopt,
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absl::nullopt));
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GetVideoSendStream()->SetSource(
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&frame_forwarder, DegradationPreference::MAINTAIN_FRAMERATE);
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test::FrameGeneratorInterface::VideoFrameData frame_data =
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frame_generator->NextFrame();
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VideoFrame frame = VideoFrame::Builder()
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.set_video_frame_buffer(frame_data.buffer)
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.set_update_rect(frame_data.update_rect)
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.build();
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frame_forwarder.IncomingCapturedFrame(frame);
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});
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EXPECT_TRUE(renderer.Wait())
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<< "Timed out while waiting for the frame to render.";
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SendTask(task_queue(), [this]() {
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Stop();
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DestroyStreams();
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DestroyCalls();
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});
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}
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TEST_F(CallOperationEndToEndTest, TransmitsFirstFrame) {
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class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
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public:
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void OnFrame(const VideoFrame& video_frame) override { event_.Set(); }
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bool Wait() { return event_.Wait(kDefaultTimeout); }
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rtc::Event event_;
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} renderer;
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std::unique_ptr<test::FrameGeneratorInterface> frame_generator;
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test::FrameForwarder frame_forwarder;
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std::unique_ptr<test::DirectTransport> sender_transport;
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std::unique_ptr<test::DirectTransport> receiver_transport;
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SendTask(
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task_queue(), [this, &renderer, &frame_generator, &frame_forwarder]() {
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CreateCalls();
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CreateSendTransport(BuiltInNetworkBehaviorConfig(),
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/*observer=*/nullptr);
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CreateReceiveTransport(BuiltInNetworkBehaviorConfig(),
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/*observer=*/nullptr);
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CreateSendConfig(1, 0, 0);
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CreateMatchingReceiveConfigs();
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video_receive_configs_[0].renderer = &renderer;
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CreateVideoStreams();
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Start();
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frame_generator = test::CreateSquareFrameGenerator(
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kDefaultWidth, kDefaultHeight, absl::nullopt, absl::nullopt);
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GetVideoSendStream()->SetSource(
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&frame_forwarder, DegradationPreference::MAINTAIN_FRAMERATE);
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test::FrameGeneratorInterface::VideoFrameData frame_data =
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frame_generator->NextFrame();
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VideoFrame frame = VideoFrame::Builder()
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.set_video_frame_buffer(frame_data.buffer)
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.set_update_rect(frame_data.update_rect)
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.build();
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frame_forwarder.IncomingCapturedFrame(frame);
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});
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EXPECT_TRUE(renderer.Wait())
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<< "Timed out while waiting for the frame to render.";
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SendTask(task_queue(), [this]() {
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Stop();
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DestroyStreams();
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DestroyCalls();
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});
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}
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} // namespace webrtc
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